Audio processing device, system, use and method

ABSTRACT

The invention relates to a hearing aid a cochlear implant comprising a) at least one input transducer for capturing incoming sound and for generating electric audio signals which represent frequency bands of the incoming sound, b) a sound processor which is configured to analyze and to process the electric audio signals, c) a transmitter that sends the processed electric audio signals, d) a receiver/stimulator, which receives the processed electric audio signals from the transmitter and converts the processed electric audio signals into electric pulses, e) an electrode array embedded in the cochlear comprising a number of electrodes for stimulating the cochlear nerve with said electric pulses, and f) a control unit configured to control the distribution of said electric pulses to the number of said electrodes. The control unit is configured to distribute said electric pulses to the number of said electrodes by applying one out of a plurality of different coding schemes, and wherein the applied coding scheme is selected according to characteristics of the incoming sound.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a Divisional of copending application Ser. No.16/217,964 filed Dec. 12, 2018, which claims priority under 35 U.S.C. §119(a) to Application No. 17206989.0 filed in the European Patent Officeon Dec. 13, 2017, all of which are hereby expressly incorporated byreference into the present application.

TECHNICAL FIELD

The invention relates to a hearing aid device that is configured toimplement a frequency band bundling scheme and a method of processing aninput audio signal.

BACKGROUND

A digital hearing aid device typically comprises an input transducersuch as a micro-phone to receive sound from an ambient environment andconvert the received acoustic signal to an electrical audio signal. Theelectrical audio input signal is a signal in the frequency domain and isconverted to a time-domain input signal using an analog-to-digitalconverter. Further, the time-domain input signal is converted to anumber of input frequency bands. Typically, the number of inputfrequency bands is determined by an analysis filter bank which alsoincludes filter coefficients to provide gain to selected frequencybands, e.g. according to a specific hearing situation. In a processingunit the number of frequency bands is processed in a number ofprocessing channels. The processing of the number of frequency bandsrequires sufficient computational power and consequently energy providede.g. by a battery. Eventually, the processed frequency bands areconverted via a digital-to-analog converter into an electric audiooutput signal that in turn is converted into audible sound and emittedas an acoustic output signal into an ear of a user using a(loud-)speaker, also called a receiver. The speaker can be located inthe ear canal of the user. Based on a hearing aid device as describedabove the hearing experience of a user can be improved.

Typically, a digital hearing aid device can be programmed by connectingit to an external computer. This allows additional processing featuresto be implemented. Thereby, the processing characteristics can beadjusted enabling e.g. an amplification or suppression of selectedfrequency bands. Sometimes processing characteristics can be adjusted bya user itself according to different hearing situations. Even programsthat adjust the processing characteristics automatically and adaptivelycan be implemented. Based on such programs e.g. acoustic feedback orbackground noise can be adaptively reduced or the processing of thereceived sound signal can be automatically adapted to different hearingsituations. Consequently, a user's comfort can be increased.

In order to decrease the computational effort and, thus, to save energy,it can be advantageous to bundle input frequency bands and to allocate asmaller number of frequency bands to be processed to processing channelsof a signal processing unit. After processing the smaller number offrequency bands, the processed frequency bands can be redistributed to alarger number of output frequency bands such that the resolution offrequency bands is increased again.

EP 3122072 A1 describes an audio processing device comprising an inputunit for converting a time-domain input signal to a number of inputfrequency bands and an output unit for converting a number of outputfrequency bands to a time-domain output signal. The object of thedescribed audio processing device is to provide a flexible audioprocessing scheme, e.g. adapted to characteristics of the input signal.This allows that the audio processing can be adapted to a particularacoustic environment and/or to a user's needs (e.g. hearing impairment)with a view to minimizing power consumption and/or processing frequencyresolution.

Hearing aid devices can be implemented e.g. as behind-the-ear (BTE)hearing aids which typically have the microphone arranged behind the earcanal of a user or as in-the-ear (ITE) hearing aids which typically havethe microphone arranged in the ear of a user. In both cases a speaker istypically placed inside a user's ear canal in order to stimulate theeardrum. An advantage of a BTE hearing aid is that the distance betweenmicrophone and speaker, also known as feedback path, is larger comparedto an ITE hearing aid such that the BTE hearing aid is less affected byfeedback. As a consequence, in BTE hearing aids a higher gain can beapplied to individual frequency bands without resulting in feedback.

In case a hearing aid device, e.g. a BTE or an ITE hearing aid,comprises a directional microphone with two microphones or two soundinlets, the directional system can be configured such that thedirectional pattern aims at cancelling the feedback path. This meansthat the directional response has its minimum directivity towards thefeedback path. The directional pattern represents the directionality ofthe directional system.

In U.S. Pat. No. 9,351,086 B2 an ITE hearing aid is described whichinter alia comprises a directional microphone and a feedback suppressionsystem for counteracting acoustic feedback on the basis of sound signalsdetected by the two microphones or the two sound inlets. The describedhearing aid device comprises an “open fitting” providing ventilation.The two microphones or the two sound inlets of the directionalmicrophone (forming part of a directional system) are arranged in theear canal at the same side of the receiver and sound is allowed topropagate freely between the microphones or between the inlets of thedirectional microphone and the receiver. It is preferred that thehearing aid device comprises a procedure (such as an adaptive procedure)for optimizing the directional system of the hearing aid device.Thereby, an improved feedback reduction is achieved, while allowing arelatively large gain to be applied to the incoming signal.

SUMMARY

It is an object of the invention to provide an improved hearing aiddevice.

According to the invention, the object is achieved by a hearing aiddevice comprising a first input transducer configured to receive a firstacoustic signal and to convert the first acoustic signal to a firstelectrical audio signal. The input transducer can be a microphone or anyother kind of input transducer. A microphone can be a microphoneintegrated in the hearing aid device and/or a remote microphone thatoperates wirelessly or it can be a wire-bound microphone.

A first analog-to-digital converter converts the first electrical audiosignal into a first time-domain input signal and a first input unit(comprising a first analysis filter bank) is configured to convert thefirst time-domain input signal to a number N_(I,1) of first inputfrequency bands. The number N_(I,1) of first input frequency bands isdetermined by an analysis filter bank (e.g. the first analysis filterbank). The analysis filter bank can include filter coefficients whichare configured to apply gain to selected frequency bands. Apparently, incase a frequency band has an increased likelihood of feedback to occurless gain is desirable.

The analysis filter bank could for example be a linear phase filter bankdesigned to allow distortion free combination (e.g. summation) offrequency band signals to frequency channel signals.

A first frequency band bundling and allocation unit can be configured tobundle adjacent first input frequency bands and to allocate firstfrequency bands to be processed to a number N_(P,1) of first processingchannels. The bundling can occur according to a bundling scheme that canbe a matrix which includes the information if an input frequency bandshall be bundled or not.

The advantage of bundling different frequency channels is mainly to savecomputational power as some computations are valid for a broader rangeof frequencies. Depending on the application different bundling schemeswill be optimal.

The frequency band bundling and allocation is configured to perform thebundling of input frequency bands dynamically or statically. Also amixture of static and dynamic bundling can be implemented. In case thefrequency band bundling and allocation unit is configured to perform thebundling of input frequency bands dynamically, the bundling can occurduring use of the hearing aid device. The bundling can then be adaptedto different hearing situations during hearing device operation. In caseof a static bundling of input frequency bands the frequency bandbundling and allocation unit can be programmed such that the bundlingscheme leads to predefined processing characteristics. A staticfrequency bundling scheme can be implemented e.g. by a hearing careprofessional before hearing aid device operation.

A memory unit is configured to store data indicating which of the firstN_(I,1) input frequency bands are subject to a likelihood of feedbackthat is above a (e.g. predefined) threshold. If the likelihood offeedback to occur in at least one of the input frequency bands is storedin the memory unit, the frequency bundling and allocation unit can beconfigured to determine the scheme based on the likelihood stored in thememory unit.

The likelihood of feedback to occur in at least one of the N_(I,1) inputfrequency bands can be determined based on a feedback detection unit.The feedback detection unit can be comprised in the hearing aid device.Alternatively, the feedback detection unit can be comprised in anexternal device connected e.g. wirelessly to the hearing aid device. Thelikelihood of feedback to occur (e.g. to exceed a threshold value) in aparticular frequency band can be determined in a variety of waysdescribed in the prior art (e.g. based on correlation measures, e.g.cross-correlation between input and output signals of the hearing aid).The likelihood of feedback to occur in a particular frequency band cane.g. be adaptively determined (during use of the hearing aid).Alternatively of additionally, the likelihood of feedback to occur in atleast one of the N_(I,1) input frequency bands can be determined in aprocedure in advance of use of the hearing aid by a user (e.g. duringfitting of the hearing aid to a particular user).

Accordingly, if the hearing aid device comprises the feedback detectionunit, the hearing aid device can be further configured to continuouslydetermine the likelihood of feedback to occur in at least one of theinput frequency bands. Thereby, the content of the bundling andallocation scheme will be updated continuously. The process of updatingthe bundling and allocation scheme in the dynamic case occurs accordingto the continuously determined likelihood of feedback.

In case, the likelihood of feedback to occur in at least one of theinput frequency bands can be determined dynamically, the feedbackdetection unit can be further comprised in the hearing aid device. Thefrequency band bundling and allocation unit then adjusts the bundlingscheme dynamically based on the by the feedback detection unitdetermined likelihood of feedback to occur in at least one of the inputfrequency bands.

In case a likelihood of feedback to occur in at least one of the inputfrequency bands is determined statically, a feedback detection unit isin general not comprised in the hearing aid device but is an externalfeedback detection unit. An external feedback detection can be used bye.g. a hearing care professional who uses the external feedbackdetection to detect which frequency regions or frequency bands comprisea high likelihood of feedback to occur. Accordingly, the hearing careprofessional can predefine the bundling of input frequency bands and theallocation of the input frequency bands to processing channels. In thissituation the bundling is static. To determine a static likelihood offeedback to occur in at least one of the input frequency bands, ameasurement of the feedback path can be considered. The feedback pathcan be measured by a hearing care professional e.g. during fitting ofthe hearing aid device. The measurement can be converted into a matrixwhich determines if an input frequency band is going to be bundled ornot. The matrix can be the applied bundling scheme.

The feedback path can be modeled using an adaptive filter. The output ofthe adaptive filter can be subtracted from e.g. the acoustic signalreceived by the microphone to cancel acoustic and/or mechanical feedbackpicked up by the microphone. As a consequence of feedback cancellationmore gain can be applied in the hearing aid. In general, a feedbackcancellation filter adapts on the acoustic signal received by amicrophone.

Alternatively, a semi-static scheme can be applied which can be based ona feedback measurement measured during startup of the hearing aiddevice. The semi-static scheme has the advantage that the most criticalfrequencies in terms of likelihood of feedback depend on the currenthearing aid device mounting.

A signal processing unit of the hearing aid device processes the firstfrequency bands to be processed in the number N_(P,1) of firstprocessing channels. It can be advantageous if the number N_(P,1) offirst processing channels is smaller than the number N_(I,1) of firstinput frequency bands. The processing of the input frequency bands in asmaller number of processing channels can lead to a reducedcomputational power and, thus, to the advantageous consequence of savingenergy during device operation.

In a preferred embodiment, the processing unit provides output frequencybands that correspond to the processed input frequency bands. The outputfrequency bands are combined to a digital audio output signal (e.g.using a synthesis filter bank). Using a digital-to-analog converter, thedigital audio output signal can be converted into an electrical audiooutput signal 71 that can be delivered to an output transducer.

An output transducer of the hearing aid device can be configured toconvert the electrical audio output signal into a user perceivablesignal, e.g. an acoustic output signal which, for a user, is perceivableas sound. The output transducer can be a speaker, receiver, astimulation unit of a cochlear implant are any other kind of outputtransducer.

According to the invention the first frequency band bundling andallocation unit is configured to generate a first bundling andallocation scheme which determines the bundling of the first N_(I,1)input frequency bands and the allocation of the first frequency bands tobe processed to the first N_(P,1) processing channels. The firstbundling and allocation scheme depends on the likelihood of feedback tooccur in at least one of the first N_(I,1) input frequency bands.

The inventors recognized that if a frequency band comprises a highlikelihood of feedback to occur it can be advantageous to have a highfrequency resolution in that frequency region. This means that inputfrequency bands in the frequency region are not bundled. If a frequencyband has a high likelihood of feedback to occur, it is desirable toapply only little or no gain to the respective frequency band.Consequently, feedback can be reduced or suppressed in an efficientmanner. In contrast, it might be desirable to apply higher gain tofrequency bands that are not subject to a high likelihood of feedback.Moreover, frequency bands that comprise a lower likelihood of feedbackto occur can be bundled. As a consequence, the input frequency bands canbe processed in a smaller number of frequency bands such that thecomputational effort and, thus, the energy consumption of the hearingaid device can be reduced.

The bundling and allocation scheme is stored in the memory unit and canbe adjusted dynamically or statically or semi-statically. The bundlingand allocation scheme determines if a frequency band of the number of N₁input frequency bands shall be bundled or not. Accordingly, feedback canbe counteracted in a very efficient manner by maintaining a highfrequency resolution in the frequency region of a frequency band thatcomprises a high likelihood of feedback to occur. At the same time, theenergy consumption can be reduced because in frequency regions whichcomprise frequency bands with a smaller likelihood of feedback to occurcan be bundled.

Sometimes feedback is also referred to as howl. Moreover, one could alsouse the term distance to feedback limit instead of likelihood offeedback to occur. The term distance to feedback expresses how much moregain can be applied until reaching the maximum allowable gain before thehearing aid device is too close to feedback or before the sound qualityof an acoustic output signal. The maximum allowable gain depends on themeasured feedback path and the currently applied gain.

An alternative definition of a likelihood of feedback is provided byusing the term gain margin, which is the amount of gain left beforeresulting in feedback. For example, a gain margin of e.g. 3 dB meansthat 3 dB more can be applied before resulting in feedback.

The hearing aid device may optionally further comprise a first frequencyband redistribution unit that is configured to redistribute the N_(P,1)processing channels to a number N_(O,1) of first output frequency bands.After processing the number of input frequency bands in a smaller numberof processing channels, the processed frequency bands are redistributedto a larger number of output frequency bands. Consequently, thefrequency resolution can be increased again compared to the number ofprocessing channels.

In a preferred embodiment the first bundling and allocation scheme is atwo-dimensional matrix representing the number N_(I,1) of first inputfrequency bands and the number N_(P,1) of first processing channelswherein for each of the N_(I,1) input frequency bands, thetwo-dimensional matrix includes a bundling value.

If the likelihood of feedback is above the threshold the bundling valuecan e.g. be zero and the respective input frequency band is not bundledwithin the N_(P,1) processing channels, and if the likelihood offeedback is below the threshold the bundling value can e.g. be one andthe respective input frequency band is bundled within the N_(P,1)processing channels. The likelihood of feedback to occur in at least oneof the input frequency bands can be detected e.g. either by a feedbackdetection unit that is comprised in the hearing aid device or by anexternal feedback detection unit. The threshold may be predefined, e.g.during a fitting session, or adaptively determined, during use of thehearing device. The threshold may be frequency dependent, e.g. dependenton a user's hearing profile (need for amplification), the hearing devicestyle (open or closed), etc.

The bundling value could be between zero and one.

The frequency band bundling and allocation unit may be configured todynamically adapt the number NP,1 of first processing channels and/orthe number NP,2 of second processing channels during normal use of thehearing aid device. A consequence of an adaptive bundling is that theinstrument is adaptively re-calibrated. E.g. internal level estimatesdepend on the bundling scheme. Calculating a level of a frequency band,when adding two frequency bands, then the level increases, and in orderto prevent the change in level a re-calibration of the level is needed.Alternatively, a fixed number of bundling schemes could be stored in theinstrument along with the corresponding calibrating values.

The bundling and allocation scheme can be a two-dimensional matrixcomprising ones and zeroes, where ones define a band to be bundled andzeroes defines bands not to be bundled. For example, a column can definethe processing channels N_(P),1 and a row can define input frequencybands N_(I) or vice versa. Thus, the bundling and allocation schemedetermines which of the input frequency bands shall be bundled and/orallocated based on a likelihood of feedback to occur the respectiveinput frequency band.

The signal processing unit can optionally be configured to determine afirst filter coefficient for each of the N_(I,1) input frequency bandsbased on the first bundling and allocation scheme, and wherein theacoustic output signal comprises a summation of the filter coefficientseach multiplied by the respective of the N_(O,1) output frequency bands.The filter coefficients are determined such that the feedback responsebetween the microphone and the speaker is reduced. Each filtercoefficient includes an imaginary part and a real part. The imaginarypart can be determined in a way that the feedback response is reduced asmuch as possible without affecting the inherent speech information orwith as little distortion of the inherent speech information aspossible. In case that there is more than one microphone comprised inthe hearing aid, each microphone comprises its individual set of inputfrequency bands which are bundled, allocated to processing channels andsubsequently processed in the signal processing unit.

By introducing complex filter coefficients to the filter bank makes thebundling relevant for time-domain bandpass filters.

Preferably, the number of Ni input frequency bands is the same as thenumber of No output frequency bands.

In a preferred embodiment of the invention the hearing aid devicecomprises

-   -   a second microphone configured to receive a second acoustic        signal and to convert the second acoustic signal to a second        electrical audio signal,    -   a second analog-to-digital converter for converting the second        electrical audio signal into a second time-domain input signal,    -   a second input unit comprising a second analysis filter bank        which is configured to convert the second time-domain input        signal to a number N_(I,2) of second input frequency bands        wherein the number N_(I,2) of second input frequency bands is        determined by the second analysis filter bank,    -   a second frequency band bundling and allocation unit which is        configured to bundle adjacent second input frequency bands and        to allocate second frequency bands to be processed to a number        N_(P,2) of second processing channels,

wherein the memory unit is configured to store data indicating which ofthe second N_(I,2) input frequency bands is subject to a likelihood offeedback that is above the threshold, and wherein the signal processingunit is adapted to process the second input frequency bands to beprocessed in the number N_(P,2) of second processing channels, and wherethe number N_(P,2) of second processing channels is smaller than thenumber N_(I,2) of second input frequency bands, and wherein the secondfrequency band bundling and allocation unit is configured to generate asecond bundling and allocation scheme which determines the bundling ofthe second N_(I,2) input frequency bands and the allocation of thesecond frequency bands to be processed to the second N_(P,2) processingchannels based on the likelihood of feedback to occur in at least one ofthe second N_(I,2) input frequency bands. The first and secondmicrophones create a directional system. The directional system can beconfigured such that the directional pattern aims at cancelling thefeedback path. This means that the directional response has its minimumdirectivity towards the feedback path.

The hearing aid device may further comprise a second frequency bandredistribution unit that is configured to redistribute the N_(P,2)processing channels to a number N_(O,2) of second output frequencybands. In this embodiment, the signal processing unit may be configuredto determine a second set of filter coefficients for each of the secondN_(I,2) input frequency bands based on the second bundling andallocation scheme. The first filter coefficients of the first set offilter coefficients and the second filter coefficients of the second setof filter coefficients comprise a real part and an imaginary part. Theimaginary part of the first and second filter coefficients is determinedsuch that the likelihood of feedback to occur is minimised and such thatthe impact on the part of the acoustic output signal which does notcomprise feedback is minimum. The acoustic output signal comprises asummation of the respective first filter coefficients each multiplied bythe respective of the first N_(O,1) output frequency bands and thesecond filter coefficients each multiplied by the respective of thesecond N_(O,2) output frequency bands. In other words, the hearing aiddevice may comprise a beamformer filtering unit for providing abeamformed signal (in spatially filtered frequency sub-bands) based onsaid first and second output frequency bands. The beamformed signal maybe further processed (e.g. to apply frequency dependent amplification orattenuation to the spatially filtered frequency sub-bands, e.g. tocompensate for a user's hearing impairment) or presented to a user viaelectrodes implanted in the auditory nerve, or converted to atime-domain audio output signal by a synthesis filter bank forpresentation to user via a speaker or via a vibrator of abone-conduction hearing aid.

Having two microphones, each of the microphones may be bundleddifferently if the feedback path of each microphone is not the same.

In a binaural hearing aid system, one or more microphones in eachhearing aid may have different or same bundling scheme.

If the analysis filter bank is defined in the time-domain, filtercoefficients are used. In contrast, if the analysis filter bank isdefined in the frequency domain, complex weights are used.Alternatively, the frequency resolution within a frequency band can beincreased using complex filter coefficients instead of a single complexweight.

Applying spatial filtering in the frequency bands to be processed allowsreducing feedback. For that reason, it is advantageous to have thehighest frequency resolution in frequency regions that comprise at leastone frequency band with a high likelihood of feedback. In frequencyregions with frequency bands that comprise a high likelihood offeedback, spatial filtering of the respective frequency bands is themore efficient for counteracting feedback if the frequency band isnarrow.

In a preferred embodiment, the likelihood of feedback is determined by afeedback detection unit which is comprised in the hearing aid device orwhich is comprised by an external device. In case the feedback detectionunit is comprised in the hearing aid device, it is possible todynamically update the bundling and allocation scheme according tochanging feedback situations. If the feedback detection unit is notcomprised within the hearing aid device but is configured as an externalfeedback detection unit, it is possible to install a static bundling andallocation scheme.

In a preferred embodiment, the feedback detection unit is configured todetermine the likelihood of feedback between the output of the speakerand the input of the first microphone defining a first feedback path andbetween the output of the speaker and the input of the second microphonedefining a second feedback path. If the hearing aid device comprises twomicrophones each microphone can be subject to feedback independent ofthe other. The first and second microphones receive first and secondacoustic signals that are converted to first and second electrical audiosignals, respectively. The first and second analog-to-digital convertersconvert the first and second audio signals into first and secondtime-domain input signals, respectively. Accordingly, the first andsecond input units convert the first and second time-domain inputsignals to a number of first and second frequency bands, respectively.To dynamically determine whether at least one of the first and secondinput frequency bands comprises a high likelihood of feedback, a firstand second feedback path can be determined. The memory unit stores dataindicating which of the first and second input frequency bands aresubject to a likelihood of feedback that is above a (e.g. predefined)threshold. Subsequently, the first and second frequency bundling andallocation units bundle the first and second input frequencies accordingto a first and second bundling and allocation scheme.

The feedback detection unit can optionally be configured to dynamicallydetermine the likelihood of feedback to occur in at least one of thefirst N_(I,1) input frequency bands and/or in at least one of the secondN_(I,2) input frequency bands, and wherein the first frequency bandbundling and allocation unit and/or the second frequency band bundlingand allocation unit is/are configured to dynamically control thebundling and allocation of the first N_(I,1) input frequency bandsand/or of the N_(I,2) second input frequency bands, respectively.

If the feedback situation changes over time, changing frequency bandsmay comprise a high likelihood of feedback. The bundling and allocationscheme can be dynamically adjusted to these changes. Accordingly,bundling of first and second frequency bands can be dynamicallycontrolled by the first and second frequency bundling and allocationunits, respectively.

The feedback detection unit can preferably further be configured toperform adaptive feedback cancellation to counteract acoustic feedbackon the basis of acoustic signals detected by the first microphone and/orthe second microphone.

Already at the stage of the feedback detection unit feedback, feedbackcancellation can be performed using adaptive feedback cancellation.

In a preferred embodiment, the feedback detection unit is configured toadaptively track feedback path changes over time based on a linear timeinvariant filter which are adapted to estimate the first and/or thesecond feedback path wherein first and/or the second filter coefficientsare updated over time.

If feedback path changes can be tracked over time, the bundling andallocation scheme stored in the memory unit can be dynamically updated.As a result, the performance of the hearing aid device can be improvedwith respect to continuously changing hearing situations. At the sametime, frequency bands that do not comprise a high likelihood of feedbackcan be bundled. Consequently, a smaller number of frequency bands isactually processed leading to the advantageous result that thecomputational effort needed is reduced. This may lead to a reduced powerconsumption during use of the hearing aid device.

In a preferred embodiment, the frequency band bundling and allocationunit is configured to dynamically adapt the number N_(P,1) of firstprocessing channels and/or the number N_(P,2) of second processingchannels during normal use of the hearing aid device.

The dynamical adaptation of the number of first and second processingchannels during normal use of the hearing aid device leads to animproved hearing experience as the likelihood of feedback to occur in atleast one of the first and second frequency bands is continuouslytracked and reduced by applying a filter to the respective frequencyband. Moreover, due to the bundling of frequency bands having only asmall likelihood of feedback the computational power is reduced leadingto the advantage that the hearing aid device can be used for a longertime before it is necessary to change the battery.

The first and/or the second frequency band bundling and allocation unitcan be configured to allocate the first and/or the second inputfrequency bands, respectively, to the respective first and/or secondprocessing channels according to a user's hearing impairment.

In a preferred embodiment, the signal processing unit is furtherconfigured to process speech intelligibility information, and whereinthe signal processing unit is further configured to prioritize theprocessing of the first and second frequency bands to be processedeither towards cancelling noise and improving speech intelligibility ortowards cancelling feedback in frequency bands where only little speechintelligibility improvement is expected.

In a preferred embodiment of the previous embodiment, the signalprocessing unit is further configured to prioritize the processing ofthe frequency bands based on the measured first and/or second feedbackpath(s) and a speech intelligibility index.

If there exists a frequency region with a high likelihood of feedback tooccur and if that region is expected to benefit only little from noisereduction, directional processing can be applied aiming at cancellingthe feedback path. In frequency regions with a small likelihood offeedback to occur and which benefits from noise reduction, thedirectional processing can be applied aiming at improving speechintelligibility. In general, low (typically below 1000 Hz) and mediumfrequency regions contribute the most to the speech intelligibility suchthat in these frequency regions speech intelligibility can be improvedby noise reduction. Moreover, the low and medium frequency regions ingeneral comprise a lower a smaller likelihood of feedback. In contrastthe higher frequency regions typically contribute less to the overallspeech intelligibility. Consequently, for the higher frequency regionsit can be advantageous to prioritize the directional processing towardscancelling the feedback path.

In a preferred embodiment, the first or the second or both frequencyband bundling and allocation units are configured to bundle adjacentinput frequency bands and to allocate the respective frequency bands tobe processed for as few processing channels as necessary. If thefrequency bands to be processed are processed in as few processingchannels as necessary, the computational power can be reduced, andenergy can be saved. The term “as few processing channels as necessary”refers to the situation where the bundling of input frequency bands isoptimized towards efficiently counteracting feedback on the one hand andbundling not more frequency bands as necessary for efficientlycounteracting feedback. For example, if frequency bands are bundledalthough the likelihood of feedback is small for the respectivefrequency bands, the bundling was not necessary. This situation is notoptimized in the sense of the term “as new processing channels asnecessary”.

The hearing aid device can be a hearing instrument, a hearing aid, abone conduction hearing aid, a headset, an earphone, an ear protectiondevice, an active ear protection system, a handsfree telephone system, amobile telephone, a teleconferencing system, a public address system, akaraoke system, a classroom amplification systems or a combinationthereof.

The object of the invention is further achieved by a hearing aid devicesystem comprising two or more hearing aid devices according to at leastone of the previous embodiments, wherein the hearing aid devices areadapted for exchanging information about the bundling of input frequencybands, preferably via a wireless communication link.

In a preferred embodiment, the hearing aid device system can beconfigured to provide that the same bundling scheme is applied in bothhearing aid devices of a binaural system by exchanging synchronizingcontrol signals between the two hearing aid devices.

According to another aspect of the invention, the aforementioned objectis achieved by a method of processing an input audio signal comprising

-   -   receiving a first acoustic signal and converting the first        acoustic signal to a first electrical audio signal        -   converting the first electrical audio signal into a first            time-domain input signal,    -   converting the first time-domain input signal to a number        N_(I,1) of first input frequency bands wherein the number        N_(I,1) of first input frequency bands is determined by a first        analysis filter bank,    -   bundling of adjacent first input frequency bands and allocating        first frequency bands to be processed to a number N_(P,1) of        first processing channels,    -   storing data indicating which of the first N_(I,1) input        frequency bands are subject to a likelihood of feedback that is        above a (e.g. predefined) threshold,    -   generating a first bundling and allocation scheme which        determines the bundling of the N_(I,1) input frequency bands and        the allocation of the first frequency bands to be processed to        the first N_(P,1) processing channels wherein said first        bundling and allocation scheme depends on the likelihood of        feedback to occur in at least one of the N_(I,1) input frequency        bands,    -   processing the first frequency bands to be processed in the        number N_(P,1) of first processing channels, wherein the number        N_(P,1) of first processing channels is smaller than the number        N_(I,1) of first input frequency bands,    -   determining first filter coefficients for each of the N_(I,1)        input frequency bands based on the first bundling and allocation        scheme,    -   redistributing the N_(P,1) processing channels to a number        N_(O,1) of first output frequency bands, and    -   emitting an acoustic output signal into an ear of a user,        wherein the acoustic output signal comprises a summation of the        first filter coefficients each multiplied by the respective of        the N_(O,1) output frequency bands.

The summation may be replaced by a linear combination.

In a preferred embodiment of the aforementioned aspect, the method ofprocessing an input audio signal further comprises

-   -   receiving a second acoustic signal and converting the second        acoustic signal to a second electrical audio signal    -   converting the second electrical audio signal into a second        time-domain input signal,    -   converting the second time-domain input signal to a number        N_(I,2) of second input frequency bands wherein the number        N_(I,2) of second input frequency bands is determined by a        second analysis filter bank,    -   bundling of adjacent second input frequency bands and allocating        second frequency bands to be processed to a number N_(P,2) of        second processing channels,    -   storing data indicating which of the second N_(I,2) input        frequency bands are subject to a likelihood of feedback that is        above a (e.g. predefined) threshold,    -   generating a second bundling and allocation scheme which        determines the bundling of the N_(I,2) input frequency bands and        the allocation of the second frequency bands to be processed to        the second N_(P,2) processing channels wherein said second        bundling and allocation scheme depends on the likelihood of        feedback to occur in at least one of the N_(I,2) input frequency        bands,    -   processing the second frequency bands to be processed in the        number N_(P,2) of second processing channels, wherein the number        N_(P,2) of second processing channels is smaller than the number        N_(I,2) of second input frequency bands,    -   determining second filter coefficients for each of the N_(I,2)        input frequency bands based on said second bundling and        allocation scheme,    -   redistributing the N_(P,2) processing channels to a number        N_(O,2) of second output frequency bands, and    -   emitting an acoustic output signal into an ear of a user,        wherein the acoustic output signal comprises a summation of the        first filter coefficients each multiplied by the respective of        the first N_(O,1) output frequency bands and the second filter        coefficients each multiplied by the respective second of the        N_(O,2) output frequency bands.

The number N_(P,2) of second processing channels may be the same as thenumber N_(P,1) of first processing channels.

The likelihood of feedback may depend on the measured feedback path toeach of the microphones, as it is desirable that the same bundlingscheme is applied to each microphone. Otherwise, it becomes difficult tocombine the two microphone signals.

In a preferred embodiment, a data processing system comprises aprocessor and program code means, adapted to cause the processor toperform the steps of the method of at least one of the twoaforementioned aspects.

A cochlear implant may comprise

-   -   at least one input transducer for capturing incoming sound and        for generating electric audio signals which represent frequency        bands of the incoming sound,    -   a sound processor which is configured to analyze and to process        the electric audio signals,    -   a transmitter that sends the processed electric audio signals,    -   a receiver/stimulator, which receives the processed electric        audio signals from the transmitter and converts the processed        electric audio signals into electric pulses,    -   an electrode array embedded in the cochlear comprising a number        of electrodes for stimulating the cochlear nerve with said        electric pulses,    -   a control unit configured to control the distribution of said        electric pulses to the number of said electrodes.

In the cochlear implant as disclosed above the distribution of saidelectric pulses to the number of said electrodes is performed byapplying one out of a plurality of different coding schemes wherein theapplied coding scheme is selected according to characteristics of theincoming sound.

Also for cochlear implants, it can be the case that there are stimuli,where it is known that some frequency regions are not used. An exampleis a telephone conversation, where the signal is band-limited up toaround 3500 Hz. Given this information, one could encode the electrodesavailable according to a specific hearing situation. Using the exampleof a telephone conversation, the telephone signal could be distributedto a number of electrodes in a different way than in other hearingsituations.

For example, it could be beneficial to use all available electrodes orto increase the stimulation rate in case that not all frequencies needto be stimulated. However, an adaptation to different hearing situationsrequires that different coding schemes can be applied. It may also bereasonable to apply a stimuli-specific coding scheme for listening tomusic.

In the cochlear implant the sound processor can optionally be configuredto analyze the characteristics of the incoming sound.

In a preferred embodiment of the cochlear implant, the control unit isconfigured to distribute the electric pulses to the number of electrodesaccording to a coding scheme for a telephone conversation and/oraccording to a coding scheme for listening to music and/or according tofurther coding schemes.

In a preferred embodiment of the cochlear implant, the coding scheme forlistening to music is configured such that high frequency channelsconvey rhythm and low frequency channels resolve tonal information.Typically, currently used coding schemes are optimized towardsunderstanding speech. This requires that a lot of information is encodedin the envelope. However, one could imagine encoding music informationin a different way e.g. by using high frequency bands to convey rhythmrather than spreading it across all bands and low frequency bands toresolve tonal information.

The sound processor in the cochlear implant can optionally be configuredto analyze the electric audio signals which represent frequency bands ofthe incoming sound with respect to an information content and to processonly frequency bands that contain meaningful information such that asmaller number of electrodes than the total number of electrodesavailable is used for stimulating the cochlear nerve.

In a preferred embodiment of the cochlear implant the audio processingdevice is configured to activate a power saving mode in which theincoming sound is analyzed by the sound processor and only frequencybands of the incoming sound that contain meaningful information aretransmitted to the electrodes. In order to reduce power consumption ofthe cochlear implant, some channels of the cochlear implant could beturned off depending on an input channel. If an acoustic input signalcontains reduced or only little information above e.g. 3 kHz, theprocessing above 3 kHz could be turned off in order to save power.Accordingly, a smaller number of electrodes could be used forstimulating the cochlear nerve. Alternatively, if the battery of thecochlear implant is getting low, a special power saving mode could beactivated, in which the acoustic input signal is analysed and onlyfrequency bands that contain a certain information content (i.e.modulated signals) are delivered to the electrodes.

In a preferred embodiment of the cochlear implant the power saving modeis configured to use preferably 1-2 broad frequency bands which in casethat the incoming sound is above a predefined amplitude threshold aretransmitted to preferably 1-2 electrodes to convey a modulation forsound awareness. The described scenario refers to an extreme powersaving mode in which only 1-2 broad bands are used and mapped to 1-2electrodes to convey a modulation for sound awareness and only if thereceived acoustic input signal is above a predefined threshold level.

The entering of the cochlear implant into the power saving mode may bedepended on a user's interaction or reaction to an incoming sound to theone or more microphones, such as head movement or a reply captured bythe microphone(s).

The control unit can optionally be configured to control thedistribution of electric pulses to the number of electrodes such thatelectric pulses are delivered to at least every second electrode inorder to reduce frequency channel interactions. By simulating on everysecond electrode only channel interactions can be reduced. As aconsequence, in this stimulation mode a user needs to adapt to aspecific frequency map that is different to a commonly used program.

In a preferred embodiment of the cochlear implant, at least one wallchannel is provided to reduce channel interactions, wherein the wallchannel is a channel in which no signal is presented and which isadjacent to the edge of a channel in which a signal is presented. Inorder to give an increased band-limited auditory nerve response, aso-called wall channel can be introduced which can be adjacent to theedge of the band in which a respective signal is presented. For example,the wall channel could be the next band above 3.5 kHz when a user is ina telephone conversation. The idea behind this is to inhibit a spread ofexcitation into high-frequency regions which, lacking their ownstimulus, might respond more readily to stimulation from lower-frequencyelectrodes. In essence, one may find that high-frequency auditory nervefibres encode a highly degraded version of an edge frequency band. Thismight be confusing or distracting for a user.

In a preferred embodiment of the cochlear implant, a wall channelstimulus within the wall channel is a low-level pulse, preferably asub-threshold pulse or a supra-threshold pulse. As a consequence of awall channel being a low-level pulse, a response adjacent to thepassband is created that is low enough in level to be of little or of noperceptual relevance and that occupies respective neurons, such thatthey do not respond much to spread of excitation from lower-frequencyelectrodes.

Two or more cochlear implants according to at least one ofaforementioned embodiments of the cochlear implant can also be comprisedin a cochlear implant system, wherein the cochlear implants can beadapted for exchanging information about the applied coding scheme.Preferably the exchange of information is provided via a wirelesscommunication link.

In a preferred embodiment of the cochlear implant system the cochlearimplant system can be configured to provide that the same coding schemeis applied in both cochlear implants of a binaural system by exchangingsynchronizing control signals between the two cochlear implants.

BRIEF DESCRIPTION OF DRAWINGS

The objects of the disclosure may be best understood from the followingdetailed description taken in conjunction with the accompanying figures.The figures are schematic and simplified for clarity, and they just showdetails to improve the understanding of the claims, while other detailsare left out. Throughout, the same reference numerals are used foridentical or corresponding parts. The individual features of each objectmay each be combined with any or all features of the other objects.These and other objects, features and/or technical effect will beapparent from and elucidated with reference to the illustrationsdescribed hereinafter in which:

FIG. 1: shows a sketch of a hearing aid device;

FIG. 2: schematically shows the processing steps of an acoustic signalreceived by one microphone;

FIG. 3: schematically shows the processing steps of acoustic signalsreceived by two microphones;

FIG. 4: illustrates the feedback path and the directional pattern of abehind-the-ear (BTE) hearing aid device;

FIG. 5: illustrates the basic principle of an in-the-ear (ITE)microphone;

FIG. 6: exemplary depicts frequency bundling according to prior art;

FIG. 7: exemplary depicts frequency bundling according to a likelihoodof feedback to occur in respective frequency bands;

FIG. 8: exemplary depicts a prioritization of the processing offrequency bands to be processed either towards cancelling noise andimproving speech intelligibility or towards cancelling feedback;

FIG. 9: shows a sketch of a cochlear implant;

FIG. 10: illustrates a bundling of two matrixes,

FIG. 11; shows how the feedback path depends on the position of aspeaker in an ear of a user,

FIG. 12; illustrates how an input signal with a limited bandwidth (suchas a telephone signal) can be distributed to bands covering a widerfrequency range.

DETAILED DESCRIPTION

The detailed description set forth below in connection with the appendeddrawings is intended as a description of various configurations. Thedetailed description includes specific details for the purpose ofproviding a thorough understanding of various concepts. However, it willbe apparent to those skilled in the art that these concepts may bepracticed without these specific details. Several object of the hearingdevice system and methods are described by various blocks, functionalunits, modules, components, circuits, steps, processes, algorithms, etc.(collectively referred to as “elements”). Depending upon particularapplication, design constraints or other reasons, these elements may beimplemented using electronic hardware, computer program, or anycombination thereof.

A hearing device may include a hearing aid that is adapted to improve oraugment the hearing capability of a user by receiving an acoustic signalfrom a user's surroundings, generating a corresponding audio signal,possibly modifying the audio signal and providing the possibly modifiedaudio signal as an audible signal to at least one of the user's ears.The “hearing device” may further refer to a device such as an earphoneor a headset adapted to receive an audio signal electronically, possiblymodifying the audio signal and providing the possibly modified audiosignals as an audible signal to at least one of the user's ears. Suchaudible signals may be provided in the form of an acoustic signalradiated into the user's outer ear, or an acoustic signal transferred asmechanical vibrations to the user's inner ears through bone structure ofthe user's head and/or through parts of middle ear of the user orelectric signals transferred directly or indirectly to cochlear nerveand/or to auditory cortex of the user.

The hearing device is adapted to be worn in any known way. This mayinclude i) arranging a unit of the hearing device behind the ear with atube leading air-borne acoustic signals or with a receiver/loudspeakerarranged close to or in the ear canal such as in a Behind-the-Ear typehearing aid or a Receiver-in-the Ear type hearing aid, and/or ii)arranging the hearing device entirely or partly in the pinna and/or inthe ear canal of the user such as in a In-the-Ear type hearing aid orIn-the-Canal/Completely-in-Canal type hearing aid, or iii) arranging aunit of the hearing device attached to a fixture implanted into theskull bone such as in Bone Anchored Hearing Aid or Cochlear Implant, oriv) arranging a unit of the hearing device as an entirely or partlyimplanted unit such as in Bone Anchored Hearing Aid or Cochlear Implant.

A hearing device may be part of a “hearing system”, which refers to asystem comprising one or two hearing devices, disclosed in presentdescription, and a “binaural hearing system” refers to a systemcomprising two hearing devices where the devices are adapted tocooperatively provide audible signals to both of the user's ears. Thehearing system or binaural hearing system may further include auxiliarydevice(s) that communicates with at least one hearing device, theauxiliary device affecting the operation of the hearing devices and/orbenefitting from the functioning of the hearing devices. A wired orwireless communication link between the at least one hearing device andthe auxiliary device is established that allows for exchanginginformation (e.g. control and status signals, possibly audio signals)between the at least one hearing device and the auxiliary device. Suchauxiliary devices may include at least one of remote controls, remotemicrophones, audio gateway devices, mobile phones, public-addresssystems, car audio systems or music players or a combination thereof.The audio gateway is adapted to receive a multitude of audio signalssuch as from an entertainment device like a TV or a music player, atelephone apparatus like a mobile telephone or a computer, a PC. Theaudio gateway is further adapted to select and/or combine an appropriateone of the received audio signals (or combination of signals) fortransmission to the at least one hearing device. The remote control isadapted to control functionality and operation of the at least onehearing devices. The function of the remote control may be implementedin a SmartPhone or other electronic device, the SmartPhone/electronicdevice possibly running an application that controls functionality ofthe at least one hearing device.

In general, a hearing device includes i) an input section such as amicrophone for receiving an acoustic signal from a user's surroundingsand providing a corresponding input audio signal, and/or ii) a receivingunit for electronically receiving an input audio signal. The hearingdevice further includes a signal processing unit for processing theinput audio signal and an output unit for providing an audible signal tothe user in dependence on the processed audio signal.

The input section may include multiple input microphones, e.g. forproviding direction-dependent audio signal processing. Such directionalmicrophone system is adapted to enhance a target acoustic source among amultitude of acoustic sources in the user's environment. In one object,the directional system is adapted to detect (such as adaptively detect)from which direction a particular part of the microphone signaloriginates. This may be achieved by using conventionally known methods.The signal processing unit may include amplifier that is adapted toapply a frequency dependent gain to the input audio signal. The signalprocessing unit may further be adapted to provide other relevantfunctionality such as compression, noise reduction, etc. The output unitmay include an output transducer such as a loudspeaker/receiver forproviding an air-borne acoustic signal transcutaneously orpercutaneously to the skull bone or a vibrator for providing astructure-borne or liquid-borne acoustic signal. In some hearingdevices, the output unit may include one or more output electrodes forproviding the electric signals such as in a Cochlear Implant.

It should be appreciated that reference throughout this specification to“one embodiment” or “an embodiment” or “an object” or features includedas “may” means that a particular feature, structure or characteristicdescribed in connection with the embodiment is included in at least oneembodiment of the disclosure. Furthermore, the particular features,structures or characteristics may be combined as suitable in one or moreembodiments of the disclosure. The previous description is provided toenable any person skilled in the art to practice the various objectsdescribed herein. Various modifications to these objects will be readilyapparent to those skilled in the art, and the generic principles definedherein may be applied to other objects.

The claims are not intended to be limited to the objects shown herein,but is to be accorded the full scope consistent with the language of theclaims, wherein reference to an element in the singular is not intendedto mean “one and only one” unless specifically so stated, but rather“one or more.” Unless specifically stated otherwise, the term “some”refers to one or more.

Accordingly, the scope should be judged in terms of the claims thatfollows.

FIG. 1 shows a sketch of a hearing aid device comprising a firstmicrophone 10 configured to receive a first acoustic signal 1 and asecond microphone 110 configured to receive a second acoustic signal101. In an optional embodiment of FIG. 1 that is not shown, a first andsecond microphone is located behind the ear of a user as part of abehind-the-ear (BTE) hearing aid device. In another embodiment offigurel that is not shown, a first and second microphone is located inthe ear canal of a user as part of an in-the-ear (ITE) hearing aiddevice. The first microphone 10 and the second microphone 110 convertthe first acoustic signal 1 and the second acoustic signal 101 to afirst electrical audio input signal 11 and a second electrical audioinput signal 111, respectively.

The hearing aid device further comprises a first analog-to-digital 20for converting the first electrical audio input signal 11 into a firsttime-domain input signal 21 and a second analog-to-digital converter 120for converting the second electrical audio input signal 111 into asecond time-domain input signal 121. The first 21 and second 121time-domain signals are subsequently delivered to a digital signalprocessing unit 90A. The digital signal processing unit 90A comprises afirst input unit 30 and a second input unit 130. The first input unit 30is configured to convert the first time-domain input 21 signal to anumber N_(I,1) of first input frequency bands 31. Thereby, the numberN_(I,1) of first input frequency bands 31 is determined by a firstanalysis filter bank that is comprised in the first input unit 30. Thesecond input unit 130 is configured to convert the second time-domaininput 121 signal to a number N_(I,2) of second input frequency bands131. Thereby, the number N_(I,2) of second input frequency bands 131 isdetermined by a second analysis filter bank that is comprised in thesecond input unit 130.

The hearing aid device further comprises first and second frequency bandbundling and allocation units 40, 140. The first frequency band bundlingand allocation unit 40 is configured to bundle adjacent first inputfrequency bands 31 and to allocate first frequency bands to be processed41 to a number N_(P,1) of first processing channels 51. The secondfrequency band bundling and allocation unit 140 is configured to bundleadjacent second input frequency bands 131 and to allocate secondfrequency bands to be processed 141 to a number N_(P,2) of secondprocessing channels 151.

The bundling of first input frequency bands 31 and second inputfrequency bands 131 can be based and a first bundling scheme and asecond bundling scheme that are created based on data stored in thememory 200. The data indicate which of the first N_(I,1) input frequencybands 31 and which of the second N_(I,2) input frequency bands 131 aresubject to a likelihood of feedback that is above a predefinedthreshold. In a preferred embodiment of FIG. 1 that is not shown, ahearing aid device comprises two memory units that either store dataindicating which of the first input frequency bands or which of thesecond input frequency bands are subject to a likelihood of feedbackthat is above a predefined threshold.

The first frequency bands to be processed 41 and the second frequencybands to be processed 141 are delivered to a signal processing unit 50.The signal processing unit 50 is configured to process the firstfrequency bands to be processed 41 in the number N_(P,1) of firstprocessing channels 51 and to process the second frequency bands to beprocessed 141 in the number N_(P,2) of second processing channels 151.Here it is preferred that the number N_(P,1) of first processingchannels 51 is smaller than the number N_(I,1) of first input frequencybands 31, and that the number N_(P,2) of second processing channels 151is smaller than the number N_(I,2) of second input frequency bands 131.The number of first and second processing channels, N_(P,1), N_(P,2),may be equal or different. The processing of the input frequency bandsin a smaller number of processing channels can lead to the advantage,that the computational power can be reduced. A reduced computationalpower can lead to the advantage that the power consumption of a hearingaid device can be reduced, or the limited number of NP frequency bandscan be used in the most efficient way.

The hearing aid device 100 further comprises a first frequency bandredistribution unit 60 and a second frequency band redistribution unit160. The first frequency band redistribution unit 60 is configured toredistribute the N_(P,1) processing channels 51 to a number N_(O,1) offirst output frequency bands 61 and the second frequency bandredistribution unit 160 is configured to redistribute the N_(P,2)processing channels 151 to a number N_(O,2) of second output frequencybands 161. Thereby, the number N_(O,1) of first output frequency bands61 can be larger than the number N_(P,1) of first processing channels 51and the number N_(O,2) of second output frequency bands 161 can belarger the number N_(P,2) of second processing channels. The number offirst and second output frequency bands, N_(O,1), N_(O,2), may be equalor different.

The first output frequency bands 61 and the second output frequencybands 161 are delivered to a signal combination unit 90B, where thefirst and second frequency bands are combined (e.g. on a frequency bandlevel (e.g. by forming a (possibly weighted) sum of the first and secondoutput frequency bands) and converted (e.g. by a synthesis filter bank)to a digital audio output signal 91 (in the time-domain) and deliveredto a digital-to-analog converter 70.

In an embodiment, the signal combination unit 90B comprises a beamformerfiltering unit, and/or a synthesis filter bank providing a resultingspatially filtered signal by applying (possibly) complex (frequencydependent) beamformer weights to the respective first and secondelectric audio signals. The beamformer filtering unit may e.g. beconfigured to provide a beamformer that is minimally sensitive in adirection towards the origin of feedback (the speaker) in frequencyregions where feedback is likely to occur (using a higher frequencyresolution in this frequency region according to the present disclosure)and to (e.g. adaptively) minimize (other) noise in other frequencyregions. Alternatively, all frequency bands may be directed to feedbackcancellation (e.g. always, or in situations where feedback is estimatedto be present, e.g. severe). In an embodiment, the beamformer filteringunit may be configured to cancel feedback (echo) in a low frequencyregion, e.g. below 1 kHz (e.g. in a specific echo cancelling mode, e.g.in a telephone mode, where sound is picked up by the hearing device andtransmitted to a far end listener and where sound from the far endlistener is received by the hearing device).

Using a digital-to-analog converter 70, the digital audio output signal91 is converted into an (analog) electrical audio output signal 71 thatis delivered to a speaker 80. The speaker 80 is configured to transmitan acoustic output signal 81 that is based on an electrical audio outputsignal 71 into an ear of a user of the hearing aid device 100. In apreferred embodiment of FIG. 1 that is not shown, a speaker is placed inthe ear canal of a user.

In a preferred embodiment of figurel that is not shown, units of thesame kind such as a first and a second input unit are comprised in asingle unit having the same functionality as the two separated units. Ina preferred embodiment of figurel that is not shown, a number of unitswith different functionality such as e.g. an input unit and ananalog-to-digital converter can be comprised in the same unit thatperforms the functionality of the comprised individual units. In analternative embodiment of figurel that is not shown, only one microphoneis comprised such that either the upper branch or the lower branch shownin FIG. 1 are comprised in the alternative embodiment. The individualbranch of the alternative embodiment of FIG. 1 still comprises a memoryunit with the functionality of the memory unit 200 shown in FIG. 1.

As stated above, the memory unit is configured to store data indicatingwhich of the first N_(I,1) input frequency bands and second N_(I,2)input frequency bands are subject to a likelihood of feedback that isabove a predefined threshold. Moreover, the likelihood of feedback isstored in a first and second bundling scheme that can be atwo-dimensional matrix indicating if a first and/or a second inputfrequency band shall be bundled or not. This allows to implement abundling scheme yielding that the frequency resolution in frequencyregions comprising frequency bands with a high likelihood of feedback islarger compared to frequency regions that comprise frequency bands witha smaller likelihood of feedback to occur. If the frequency resolutionin frequency regions is high, it is possible to reduce or counteract thefeedback in the respective frequency bands very efficiently. This is dueto the fact that the respective frequency bands can be selected andprocessed individually and a filter be exclusively applied to theserespective frequency bands. Moreover, frequency bands with a smalllikelihood of feedback to occur can be bundled such that thecomputational effort and thus the power consumption of the hearing aidcan be reduced.

The likelihood of feedback to occur in at least one of the first and/orsecond frequency bands can be determined by a feedback detection unit250. The feedback detection unit 250 detects the likelihood of feedbackby e.g. dynamically tracking changes in the feedback path 251. In theembodiment of FIG. 1, (only) the feedback path between the speaker 80and the second microphone 110 is estimated by feedback detection unit250. If the detected likelihood of feedback to occur in at least one ofthe second input frequency bands 131 is high, the respective informationis delivered 252 to the memory unit 200 and stored as data indicatingwhich of the second N_(I,2) input frequency bands are subject to alikelihood of feedback that is above a predefined threshold.

In an alternative embodiment that is not shown in FIG. 1, two feedbackdetection units can be comprised each working exclusively for oneprocessing branch, only. If one or more feedback detection units arecomprised in a hearing aid device, a dynamical tracking of feedback pathchanges can be implemented such that a first and/or a second bundlingscheme can be updated during hearing aid device operation according tochanging feedback situations. A first feedback detection unit can beconfigured to track a first feedback path between a first microphone anda speaker and a second feedback detection unit can be configured totrack feedback path changes in the feedback path between the secondmicrophone and the speaker.

In an alternative embodiment that is not shown in FIG. 1, the feedbackdetection unit is implemented as an external feedback detection unit. Ahearing care professional can detect frequency bands that comprise ahigh likelihood and store data indicating which of the input frequencybands are subject to a likelihood of feedback that is above a predefinedthreshold on a memory unit comprised in a hearing aid device. As aresult, a frequency band bundling scheme will be static.

The signal processing as described above can also be implemented in ahearing aid implant such as a cochlear implant. In this case, theprocessed signal would not be converted to an acoustic output signalthat is emitted by a speaker but a processed electric audio signal couldbe converted into electric pulses. Then, an electrode array comprising anumber of electrodes which are embedded in the cochlear of a user couldbe used for stimulating the cochlear nerve with said electric pulses.

FIGS. 2 and 3 which are explained in the following, focus on thedifferent signal processing steps as implemented in a hearing aid deviceaccording to a preferred embodiment of the invention. FIG. 2 focuses ona hearing aid device comprising one microphone and FIG. 3 describes ahearing aid device comprising two microphones.

FIG. 2 depicts the signal processing steps in hearing aid deviceaccording to a preferred embodiment of the invention. First, amicrophone 10 receives an acoustic signal 1 which is converted into anumber of N_(I) input frequency bands 31 (e.g. by an analysis filterbank, cf. e.g. 30 in FIG. 1). The input frequency bands 31 can bebundled or partly bundled and subsequently allocated to a number N_(P)of processing channels 51 (e.g. by a frequency band bundling andallocation unit 40, as indicated in FIG. 1). The number N_(P) ofprocessing channels 51 are processed in the signal processing unit 50.Processing in the signal processing unit 50 can include thedetermination of filter coefficients or gain values (Wp) 53 for each ofthe N_(I) input frequency bands based on e.g. a bundling and allocationscheme. After signal processing, the N_(P) processing channels areredistributed to a number N_(O) of output frequency bands (cf. NP→NOunit in FIG. 2). This allows to filter selected frequency bands that aresubject to a high likelihood of feedback or to apply different gainlevels to selected frequency bands according to specific hearingsituations.

In the embodiment shown, the number N_(I) of input frequency bands andthe number N_(O) of output frequency bands is identical as indicated bythe arrow 35. Consequently, the initial frequency resolution isrehabilitated after processing the signal in a smaller number ofprocessing channels N_(P). The acoustic output signal 81 provided by thespeaker 80 comprises a ‘summation’ of the resulting frequency sub-bandsignals determined from the contents of the N_(P) processing channels(filter coefficients 53 (Wp)) subject to a frequency band redistributionunit (cf. unit 60 (or 160) in FIG. 1) and each multiplied by therespective of the N_(I) input frequency bands 35 received from inputunit 30 (providing the N_(I) input frequency bands 31). The resultingfrequency sub-band signals, output band signals 61, are converted to atime-domain output signal 71 by a synthesis filter bank (FBS) (andpossibly a DA converter) and fed to speaker 80.

The signal processing as described above can also be implemented in ahearing aid implant such as a cochlear implant. In this case, theprocessed signal would not be converted to an acoustic output signalthat is emitted by a speaker but a processed electric audio signal couldbe converted into electric pulses. Then, an electrode array comprising anumber of electrodes which are embedded in the cochlear of a user couldbe used for stimulating the cochlear nerve with said electric pulses. Inthis case, the individual band signals (e.g. N_(P) channel signalsWp1*No1, Wp2*No2, Wp3*No3, . . . , or No redistributed output bandsignals) could be presented to a different one of the electrodes of theelectrode array.

FIG. 3 depicts the signal processing steps in hearing aid deviceaccording to a preferred embodiment of the invention. In contrast toFIG. 2, a first microphone 10 and a second microphone 110 are comprisedand configured to receive a first acoustic signal 1 and a secondacoustic signal 101, respectively. The first acoustic audio signal 1 isconverted into a number N_(I,1) of first input frequency bands and thesecond acoustic audio signal 101 is converted into a number N_(I,2) ofsecond input frequency bands. The first input frequency bands 31 can bebundled according to a first bundling scheme and the second frequencybands 131 can be bundled according to a second bundling scheme.Subsequently, the number of first frequency bands to be processed isallocated (by unit N1 p) to a number N_(P,1) of first processingchannels 51 and the number of second frequency bands to be processed isallocated (by unit N2 p) to a number N_(P,2) of processing channels 151.

The number N_(P,1) of first processing channels 51 and the numberN_(P,2) of second processing channels 151 are processed in the signalprocessing unit 50. Processing in the signal processing unit 50 caninclude the determination of a set of first filter coefficients (W1 p)54 for each of the N_(I,1) first input frequency bands and thedetermination of a set of second filter coefficients (W2 p) 55 for eachof the N_(I,2) second input frequency bands based on e.g. a likelihoodof feedback in at least one of the first and second input frequencybands. After signal processing, the N_(P,1) first processing channelsand the N_(P,2) second processing channels are redistributed to a numberN_(O,1) of first output frequency bands and to a number N_(O,2) ofsecond output frequency bands, respectively (cf. unit NP1→NO, NP2→NO).Each of the number N_(O,1) of first output frequency bands and thenumber N_(O,2) of second output frequency bands can be multiplied by anindividual (possibly complex) filter coefficient that is determined bythe signal processing unit 50. This allows suppressing feedback infrequency bands comprising a high likelihood of feedback (beamforming,cf. unit WS).

The first filter coefficients of the first set of filter coefficients(W1 p) and the second filter coefficients of the second set of filtercoefficients (W2 p) may comprise a real part and an imaginary part. Thereal and imaginary part of the first and second filter coefficients canbe determined such that the likelihood of feedback to occur is minimisedand such that the impact on the part of the acoustic output signal whichdoes not comprise feedback is minimum (e.g. using beamformingtechniques). Moreover, the acoustic output signal 81 comprises a(possibly weighted) summation of the respective first filtercoefficients each multiplied by the respective of the first N_(O,1)output frequency bands and the second filter coefficients eachmultiplied by the respective of the second N_(O,2) output frequencybands. The output frequency bands may be received (35A and 35B) from thefirst input frequency bands 31 and the second input frequency bands 131,respectively. The resulting frequency output bands 61 may be translatedto the time-domain (signal 71) by a synthesis filter bank FBS (andpossibly converted to an analog signal by DA converter) beforepresentation to the speaker 80.

The filter coefficients could have different purpose depending on theamount of feedback in the feedback path 250: In frequency bands withhigh risk of feedback, the coefficients are adapted towards minimizingfeedback. In bands, where the risk of feedback is small (e.g. dependingon a feedback path measurement, e.g. at low frequencies), thecoefficients could be adapted towards minimizing external noise. Incertain application scenarios involving large delays from output toinput, echo cancellation can appear at relatively low frequencies. Insuch cases, the coefficients may be used to minimize echo in lowfrequency bands, e.g. below 1.5 kHz or below 1 kHz.

The signal processing as described above can also be implemented in ahearing aid implant such as a cochlear implant. In this case, theprocessed signal would not be converted to an acoustic output signalthat is emitted by a speaker but a processed electric audio signal couldbe converted into electric pulses. Then, an electrode array comprising anumber of electrodes which are embedded in the cochlear of a user couldbe used for stimulating the cochlear nerve with said electric pulses(each e.g. representing contents of a different output channel or band).

FIG. 4 shows a behind-the-ear (BTE) microphone containing twomicrophones. In case a hearing aid device contains two microphones, thedirectional system can be adapted towards cancelling the estimatedfeedback path 251. The feedback paths are related to the distancebetween the first or the second microphone 10,110 and the speaker 80.For a given frequency band, a directivity pattern 260 can be improvedtowards cancelling the estimated feedback path 251 while keeping apreferred listening direction unaltered. The left part of FIG. 4represents front view of (a right part of the head of) a user wearing anembodiment of a (BTE-type) hearing device at the right ear, whereas theright part represents a side view of the hearing device at the rightear. The embodiment may represent a receiver in the ear (RITE) stylehearing device, where the speaker (receiver) is located in the ear canalor a BTE style hearing device, where the speaker (receiver) is locatedin the BTE-part, and where sound is propagated to the ear canal by atube between the BTE-part and the ear canal, cf. bold line. In theembodiments shown, the directional response has its minimum directivitytowards the feedback path 251. In frequency regions where feedback islikely to occur, a higher frequency resolution is desirable.Consequently, the frequency band bundling as implemented by a bundlingscheme can be performed such that the frequency resolution in frequencyregions with high likelihood of feedback is high.

At least one of the microphones (10, 110) may be used as a referencemicrophone for estimating feedback (cf. e.g. FIG. 1).

FIG. 5 illustrates an in-the-ear (ITE) style hearing device comprising adual microphone solution as used for feedback cancellation. A hearingaid device containing two microphones 10, 110 which are placed in theear canal of a user (cf. left part of FIG. 5) can be even better suitedfor cancelling the feedback path compared to the BTE solution presentedin FIG. 4. The two microphones 10, 110 can be spaced by about 7 to 8millimetres. Here, a fixed or an adaptive directional gain (cf.directivity pattern 260, cf. right part of FIG. 5) can be applied aimingat counteracting at the frequencies where feedback occurs.

FIG. 6 schematically depicts signal processing according to prior art(cf. e.g. EP2503794A1). A time-domain input signal is converted to anumber of input frequency bands. The number of input frequency bands isdetermined by an analysis filter bank. In the example shown thefrequency increases from bottom to top such that at the bottom part ofthe analysis filter bank low frequencies are shown and at the top parthigh frequencies are shown. In order to reduce the computational effortand thus to save energy, some input frequency bands are merged into afewer number of processing channels. After processing, the processedfrequency bands are redistributed to the initial number of inputfrequency bands. A (e.g. fixed) gain that could be complex is calculatedduring the processing and applied to the number of output frequencybands. Subsequently, the bands are processed via a synthesis filter bankin order to obtain a modified time-domain signal.

The signal processing as described above can also be implemented in ahearing aid implant such as a cochlear implant. Then the bundling offrequency bands could be used and applied to the distribution ofelectric pulses to a number of said electrodes. The distribution ofelectric pulses could e.g. be performed by applying one out of aplurality of different coding schemes and the applied coding schemecould be selected according to characteristics of an incoming sound.

FIG. 7 schematically depicts signal processing according to a preferredembodiment of the invention. A first and a second acoustic signalreceived by a first and a second microphone are converted to a first andsecond (e.g. digitized) time-domain input signal, respectively, anddelivered to an analysis filter bank (comprising first and secondanalysis filter banks for providing frequency sub-band versions of thefirst and second time-domain input signals, respectively). The frequencybands embraced by a curly bracket 300 represent frequency bands in aregion with a high likelihood of feedback. As recognized by theinventors, in regions where a likelihood of feedback is high it isdesirable to have a high frequency resolution in order to efficientlycounteract feedback. Consequently, the frequency bands where feedback islikely to occur are not bundled and the frequency bundling is performedtowards an improved cancelling of feedback. In frequency regions, wherefeedback is likely to occur, the directional system can be adaptedtowards cancelling the feedback path. In those regions which typicallyare the higher frequencies regions it is desirable to have a highfrequency resolution.

After processing (of each of the first and second microphone signals) ina smaller number of processing channels, the processed frequencychannels are redistributed to a number of output frequency bands thatcan be an identical number to the initial number of input frequencybands. During processing, filter coefficients (e.g. respective channelspecific values) are determined and subsequently applied to each of theinput frequency bands of the first and second microphone signals (cf.dashed arrow from input bands to multiplication units of eachre-distributed band). In the respective multiplication units, thedetermined filter coefficients for each frequency band of the first andsecond microphone signals are mixed with the contents of each of thecorresponding input frequency bands of the respective first and secondmicrophone signals to provide first and second output frequency bands.The unit denoted ‘+’ represents a combination of the first and secondoutput frequency bands. The unit ‘+’ may e.g. implement a weighted sumof the first and second output bands, e.g. to implement specificfrequency (band) specific beam patterns. Subsequently, the resultingfrequency sub-bands are processed via a synthesis filter bank in orderto obtain a modified time-domain signal.

The signal processing as described above can also be implemented in ahearing aid implant such as a cochlear implant. Then the bundling offrequency bands could be used and applied to the distribution ofelectric pulses to a number of said electrodes. The distribution ofelectric pulses could e.g. be performed by applying one out of aplurality of different coding schemes and the applied coding schemecould be selected according to characteristics of an incoming sound.

FIG. 8 shows a prioritization scheme for prioritizing the processing offrequency bands either towards cancelling noise and improving speechintelligibility or towards cancelling feedback. A first and a secondacoustic signal received by a first and a second microphone areconverted to a first and second (e.g. digitized) time-domain inputsignal, respectively, and delivered to an analysis filter bank.

In general, the directional processing in different frequency bandscould be prioritized either towards cancelling noise and improvingspeech intelligibility or be prioritized towards cancelling feedback infrequency regions, where only a little speech intelligibilityimprovement is achieved. Such a prioritization could be based on themeasured feedback path and the speech intelligibility band importanceindex. In order to minimize the power consumption, the bundling offrequency bands can be optimized for as few processing channels asnecessary to maintain a sufficient frequency resolution for providingthe information contained in the signal in an adequate manner.

In the low and medium frequencies (indicated by curly bracket) 400,directional processing used for noise reduction improves speechintelligibility significantly. Also, at the low frequency regions whichare typically below 1000 Hz, feedback is not likely to occur. In thehigher frequency region (indicated by curly bracket) 440, whichcontributes only a little to the overall speech intelligibility, it canbe reasonable to prioritize the directional processing from thosefrequency regions to cancel the feedback path.

In a binaural hearing aid system, the bundling scheme may be the samefor both left and right hearing aid. As a consequence, the bundlingscheme depends on the feedback path measures at both hearing aids. Inanother example, the bundling scheme may be different in the left andthe right hearing aid. In a yet another example, the bundling scheme ispartly the same at left and the right hearing aid, e.g. the bundlingscheme may be the same within a frequency range and different withinanother frequency range.

The prioritization scheme as described above can also be implemented ina hearing aid implant such as a cochlear implant. Then theprioritization of frequency bands could be used and applied to thedistribution of electric pulses to a number of said electrodes.

FIG. 9 shows a cochlear implant 1000 that comprises an external part1100 and an implanted part 1200. The external part (1100, 1010)comprises at least one input transducer 1010 for capturing incomingsound 1011 and for generating electric audio signals 1012 whichrepresent frequency bands of the incoming sound 1011. A sound processor1020 is configured to analyze and to process the electric audio signals1012 and a transmitter 1030 sends the processed electric audio signals1021 to a receiver/stimulator 1040, e.g. via an inductive link 1031. Thereceiver/stimulator 1040 receives the processed electric audio signals1021 from the transmitter 1030 and converts the processed electric audiosignals 1021 into electric pulses 1041. A control unit 1060 isconfigured to control the distribution 1061 of said electric pulses 1041to the number of said electrodes 1055. An electrode array 1050 which isembedded in (implanted in) cochlea comprises a number of electrodes 155for stimulating the cochlear nerve with said electric pulses 1041. Inthe cochlear implant 1000 the distribution of said electric pulses 1041to the number of said electrodes 155 is performed by applying one out ofa plurality of different coding schemes wherein the applied codingscheme is selected according to characteristics of the incoming sound1011.

FIG. 10 illustrates an example where the bundling and allocation schemeis a two-dimensional matrix comprising ones and zeroes, where ones forexample defines a band to be bundled and zeroes defines bands not to bebundled. For example, a column can define the processing channels NP,1and a row can define input frequency bands NI or vice versa. Thus, thebundling and allocation scheme determines which of the input frequencybands shall be bundled and/or allocated based on a likelihood offeedback to occur the respective input frequency band. Thetwo-dimensional matrix is multiplied by a one-dimensional matrixcomprising the number No of output frequency bands. Each of the outputfrequency bands are identical to the input frequency bands Ni, i.e. No1is equal to Ni1. The result shows a redistribution of the Np processingchannels to a number No of output frequency bands. This allows to filterselected frequency bands that are subject to a high likelihood offeedback or to apply different gain levels to selected frequency bandsaccording to specific hearing situations.

FIG. 11 illustrates different positions of the speaker 80 in an ear of auser of the hearing aid 100 (cf. upper left (A) and right (B) sketches)and the effect on the feedback path 250 as a function of frequency (cf.corresponding lower left and right graphs). It is seen that the peak inthe feedback path in dB has shifted upwards in frequency when placingthe speaker 80 more outwardly within the ear (right scenario B). Ingeneral, the feedback path changes according to the position of thespeaker 80 in the ear of a user and hereby different frequency rangeswith a high likelihood of feedback are to be expected. Hence,preferably, the frequency band bundling, e.g. the first and/or thesecond frequency band bundling, may be provided based on a measurementof the feedback path 250 in order to determine whether a change in thefeedback path 250 has occurred.

FIG. 12 illustrates how an input signal with a limited bandwidth (suchas a telephone signal) can be distributed to bands covering a widerfrequency range. This is for example applicable in a cochlear implant,where more information can be transmitted to the brain, if allelectrodes are stimulated in contrast to only stimulating electrodesthat covers the frequency range of the input stimuli. There-distribution of a narrow frequency band input signal to a wideroutput bandwidth can be interpreted as bandwidth extension.

1. A cochlear implant comprising at least one input transducer forcapturing incoming sound and for generating electric audio signals whichrepresent frequency bands of the incoming sound, a sound processor whichis configured to analyze and to process the electric audio signals, atransmitter that sends the processed electric audio signals, areceiver/stimulator, which receives the processed electric audio signalsfrom the transmitter and converts the processed electric audio signalsinto electric pulses, an electrode array embedded in the cochlearcomprising a number of electrodes for stimulating the cochlear nervewith said electric pulses, and a control unit configured to control thedistribution of said electric pulses to the number of said electrodes,wherein the control unit is configured to distribute said electricpulses to the number of said electrodes by applying one out of aplurality of different coding schemes, and wherein the applied codingscheme is selected according to characteristics of the incoming sound.2. A cochlear implant according to claim 1, wherein the sound processoris configured to analyze the characteristics of the incoming sound.
 3. Acochlear implant according to claim 1, wherein the distribution of saidelectric pulses to the number of said electrodes is performed accordingto a specific hearing situation.
 4. A cochlear implant according toclaim 1, configured to increase the stimulation rate in case that notall frequencies need to be stimulated.
 5. A cochlear implant accordingto claim 1, configured to apply a stimuli-specific coding scheme forlistening to music.
 6. A cochlear implant according to claim 5 wherein,the coding scheme for listening to music is configured such that highfrequency channels convey rhythm and low frequency channels resolvetonal information.
 7. A cochlear implant according to claim 1 whereinthe control unit is configured to distribute the electric pulses to thenumber of electrodes according to a coding scheme for a telephoneconversation and/or according to a coding scheme for listening to musicand/or according to further coding schemes.
 8. A cochlear implantaccording to claim 1 wherein the sound processor in the cochlear implantis configured to analyze the electric audio signals which representfrequency bands of the incoming sound with respect to an informationcontent and to process only frequency bands that contain meaningfulinformation such that a smaller number of electrodes than the totalnumber of electrodes available is used for stimulating the cochlearnerve.
 9. A cochlear implant according to claim 1 configured to activatea power saving mode in which the incoming sound is analyzed by the soundprocessor and only frequency bands of the incoming sound that containmeaningful information are transmitted to the electrodes.
 10. A cochlearimplant according to claim 1 wherein some channels of the cochlearimplant can be turned off depending on an input channel.
 11. A cochlearimplant according to claim 1 wherein a special power saving mode can beactivated, in which the acoustic input signal is analysed and onlyfrequency bands that contain a certain information content are deliveredto the electrodes.
 12. A cochlear implant according to claim 10 whereinthe power saving mode is configured to use only 1 or 2 broad frequencybands which in case that the incoming sound is above a predefinedamplitude threshold are transmitted to 1 or 2 electrodes to convey amodulation for sound awareness.
 13. A cochlear implant according toclaim 1 wherein the entering of the cochlear implant into the powersaving mode is dependent on a user's interaction or reaction to anincoming sound to the one or more microphones, such as head movement ora reply captured by the transducer(s).
 14. A cochlear implant accordingto claim 1 wherein the control unit is configured to control thedistribution of electric pulses to the number of electrodes such thatelectric pulses are delivered to at least every second electrode inorder to reduce frequency channel interactions.
 15. A cochlear implantaccording to claim 1 wherein at least one wall channel is provided toreduce channel interactions, wherein the wall channel is a channel inwhich no signal is presented and which is adjacent to the edge of achannel in which a signal is presented.
 16. A cochlear implant accordingto claim 15 wherein a wall channel stimulus within the wall channel is alow-level pulse, preferably a sub-threshold pulse or a supra-thresholdpulse.
 17. A cochlear implant according to claim 1 comprising anexternal part and an implanted part.
 18. A cochlear implant systemcomprising two or more cochlear implants according to claim 1, whereinthe cochlear implants are adapted for exchanging information about theapplied coding scheme.
 19. A cochlear implant system according to claim18 wherein the exchange of information is provided via a wirelesscommunication link.
 20. A cochlear implant system according to claim 18configured to provide that the same coding scheme is applied in bothcochlear implants of a binaural system by exchanging synchronizingcontrol signals between the two cochlear implants.
 21. A cochlearimplant comprising at least one input transducer for capturing incomingsound and for generating electric audio signals which representfrequency bands of the incoming sound, a sound processor which isconfigured to analyze and to process the electric audio signals, atransmitter that sends the processed electric audio signals, areceiver/stimulator, which receives the processed electric audio signalsfrom the transmitter and converts the processed electric audio signalsinto electric pulses, an electrode array embedded in the cochlearcomprising a number of electrodes for stimulating the cochlear nervewith said electric pulses, and a control unit configured to control thedistribution of said electric pulses to the number of said electrodes,wherein the control unit is configured to distribute the electric pulsesto the number of electrodes in dependence of characteristics of theincoming sound.
 22. A cochlear implant according to claim 21 wherein thecontrol unit is configured to distribute said electric pulses to thenumber of said electrodes by applying one out of a plurality ofdifferent coding schemes, and wherein the applied coding scheme isselected according to characteristics of the incoming sound